[OpenSIPS-Users] opensips and asterisk retransmits

[Iñaki Baz Castillo]

El Miércoles, 29 de Abril de 2009, Iñaki Baz Castillo escribió:
> El Miércoles, 29 de Abril de 2009, troxlinux escribió:
> > Hi Bogdan , I don't have any alias en mi opensips.cfg , the only thing
> > that I have is that when they make calls to the pstn they leave to
> > that ip port
> >
> > route[4] {
> >         rewritehostport("192.168.10.3:5070");
> >         route(1);
> > }
>
> You *should not* use the above route[4] for in-dialog ACK (after 200 OK),
> are you using it for that?

Check this (as Bogdan said) and try first to understand which the problem is. 
Later you could look for a solution:


# PHONE -> OPENSIPS
U +0.038456 192.168.10.30:5064 -> 192.168.10.3:5060
ACK sip:*981 at 192.168.10.3:5070 SIP/2.0 
Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bKb1e3a5d2b3f28c71 
Route: <sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3> 

# OPENSIPS -> OPENSIPS ?
U +0.000634 192.168.10.3:5060 -> 192.168.10.3:5060
ACK sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3 SIP/2.0 
Record-Route: <sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3>


This ACK is an in-dialog request (confirmation for 200 OK) so RURI 
***shouldn't*** be changed by the proxy, but you do change it and this is your 
error.

Most probably, as I said in othermail, you are modifying the RURI of the ACK 
while what you should do is bypass it as an in-dialog request (lose_route()).


-- 
Iñaki Baz Castillo <ibc at aliax.net>